Index: webrtc/api/call/audio_state.h |
diff --git a/webrtc/api/call/audio_state.h b/webrtc/api/call/audio_state.h |
index ac912773aa5fdeb12a59120b794c269605614050..1941cbf7bcd976c8aaabdf24709e6440d41443ac 100644 |
--- a/webrtc/api/call/audio_state.h |
+++ b/webrtc/api/call/audio_state.h |
@@ -10,6 +10,7 @@ |
#ifndef WEBRTC_API_CALL_AUDIO_STATE_H_ |
#define WEBRTC_API_CALL_AUDIO_STATE_H_ |
+#include "webrtc/api/audio/audio_mixer.h" |
#include "webrtc/base/refcount.h" |
#include "webrtc/base/scoped_ref_ptr.h" |
@@ -33,8 +34,9 @@ class AudioState : public rtc::RefCountInterface { |
// the AudioState itself. |
VoiceEngine* voice_engine = nullptr; |
- // The AudioDeviceModule associated with the Calls. |
- AudioDeviceModule* audio_device_module = nullptr; |
+ // The audio mixer connected to active receive streams. One per |
+ // AudioState. |
+ rtc::scoped_refptr<AudioMixer> audio_mixer; |
}; |
// TODO(solenberg): Replace scoped_refptr with shared_ptr once we can use it. |